[asterisk-dev] [Code Review] external sip attended transfer test

Jeff Peeler jpeeler at digium.com
Thu May 20 09:18:07 CDT 2010


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/662/#review2037
-----------------------------------------------------------



/asterisk/trunk/tests/sip_attended_transfer/run-test
<https://reviewboard.asterisk.org/r/662/#comment4321>

    repeated



/asterisk/trunk/tests/sip_attended_transfer/run-test
<https://reviewboard.asterisk.org/r/662/#comment4325>

    Instead of making the reactor calls based on time, why not schedule the next method call at the end of each call or transfer? In fact, if there was some type of information returned for success from pjsua you wouldn't need the reactor at all. Although it probably is good to schedule a timeout in case something gets stuck and is undetected.



/asterisk/trunk/tests/sip_one_legged_transfer/run-test
<https://reviewboard.asterisk.org/r/662/#comment4326>

    zombies



/asterisk/trunk/tests/sip_one_legged_transfer/run-test
<https://reviewboard.asterisk.org/r/662/#comment4327>

    same deal I think


- Jeff


On 2010-05-18 16:57:25, David Vossel wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/662/
> -----------------------------------------------------------
> 
> (Updated 2010-05-18 16:57:25)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> This is an external test to exercise the native sip method of doing attended transfers using the Replaces header.
> 
> A, B, and C are SIP endpoints.
> 
> A calls B
> A calls C
> A transfers B to C
> 
> This test is verified using the 'Status' manager action to make sure the channel's are bridged correctly after the transfer.
> 
> 
> To run this test, pjsip's pjsua app is required.  It can be found here, http://www.pjsip.org/download.htm.  After installing rename and move ..../pjsip-apps/bin/pjsua-x86-unknown-linux-gnu to /usr/sbin/pjsua. This puts pjsua into PATH, and this script expects it to be named 'pjsua'.
> 
> 
> Diffs
> -----
> 
>   /asterisk/trunk/tests/sip_attended_transfer/configs/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/sip_attended_transfer/configs/manager.conf PRE-CREATION 
>   /asterisk/trunk/tests/sip_attended_transfer/configs/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/sip_attended_transfer/run-test PRE-CREATION 
>   /asterisk/trunk/tests/sip_attended_transfer/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/sip_one_legged_transfer/configs/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/sip_one_legged_transfer/configs/manager.conf PRE-CREATION 
>   /asterisk/trunk/tests/sip_one_legged_transfer/configs/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/sip_one_legged_transfer/run-test PRE-CREATION 
>   /asterisk/trunk/tests/sip_one_legged_transfer/test-config.yaml PRE-CREATION 
> 
> Diff: https://reviewboard.asterisk.org/r/662/diff
> 
> 
> Testing
> -------
> 
> Tested with 1.4 and Trunk.  Trunk r263541 currently causes this test to segfault, but earlier revisions pass.
> 
> 
> Thanks,
> 
> David
> 
>




More information about the asterisk-dev mailing list